Network services using real-time delivery of multimedia data over a network have become ubiquitous. For example, packet-based telephony service, such as voice-over-IP (“VoIP”) telephony, typically includes the real-time delivery of voice, and other multimedia data types, such as video data, on a network using Real-Time Transport Protocol (RTP) to exchange information that controls the delivery of data. The perceived quality of VoIP telephony service can be determined by various parameters affecting the real-time delivery of the data over the network, such as network conditions, and the network resources allocated for the delivery of the data.
Various conditions along the network may adversely affect the real-time delivery of multimedia data throughout a network. For example, network congestion and the capacity of various components along the network, including the endpoints, may impact performance. In systems using VoIP technology, voice and video signals are converted into network packets, which may be transported via a variety of suitable Internet Protocol (IP) based protocols. When data is transferred via packets in accordance with the IP-based protocols, certain packets may be dropped, for example, due to network congestion at a router or link in the transmission pathway that may receive packets at a greater rate than it is capable of transmitting the packets. This packet dropping may result in a loss of information and a decreased user-perceived quality of service. While certain traffic duplicating systems have been developed to address these problems, many such systems operate based on data sourced from only the protocol level. Therefore, systems and methods are needed to address one or more of these drawbacks of conventional systems.